The present invention relates generally to a voice encode/decode subsystem, and move particularly to an adaptive Delta Modulation (ADM) technique that is used to encode and decode voice for PCM transmission.
In the field of aircraft testing the need to efficiently record the cockpit voice communication without consuming a significant amount of the acquisition frame bandwidth has been an issue for years. There are methods, based on commercially available products, that allow for voice placement into PCM streams that will satisfy the requirement of relatively low bandwidth consumption. The present invention relates to a design that makes minimal demand on bandwidth, with the freedom to vary the placement of the voice within the minor acquisition frame.
The data acquisition system this subsystem was built to integrate into is called the Airborne Test Instrumentation System (ATIS) designed and developed by SCI in Huntsville, Ala. The system consists of Format, or System Control Unit and Analog and Digital Remote Units and a Data Bus Interface Unit. Data is captured in synchronization with a system base clock called a bit rate clock (BRC) that is generated by an oscillator within the Format or System control unit. Once captured the data may be sent to a recorder, or a telemetry subsystem for transmission. There is a limited on board monitoring subsystem to sample some of the parameters in "real-time".
There are various methods to acquire voice for placement into a data acquisition PCM stream for recording and/or transmission. Methods such as analog modulations via VCOs place voice on its own track on a storage medium, and standard sample and hold code a magnitude over 2.sup.n levels. These methods either degrade quality, inefficiently use storage space, or consume severely needed channel bandwidth. It can integrate into any system that uses a request pulse for serial or parallel acquisition of TTL signals.
Of the many ways to convert an analog signal to digital, one means that allows for a relatively small bandwidth is Delta Modulation (DM). In PCM the sampled signal is quantified into L levels and transmits as n=log.sub.2 L pulses per sample. In DM we are encoding only the difference in successive signals. So, basically, DM carries information about the derivative, or slope of the signal. Therefore, we are encoding information in one pulse instead of n pulses. This is a great savings in bandwidth, but there is a trade-off with quantization error. This is overcome by sampling faster, usually 4-8 times higher than PCM. The bare bones method isn't very practical. References:
1. Taub, Herbert, 1918-Principles of Communications Systems McGraw Hill; 1986, 1971 PA1 2. Lathi, B. P. Modern Digital and Analog Communications Systems CBS College Publishing; 1983